Standards are prevalent in telephony and data communications industries. Specifying and clarifying design principles, communication processes, test procedures, and environmental conditions have helped to assure product quality and multi-vendor interoperability. The voice over Internet Protocol (IP) community has a collection of standards for voice codecs derived out of multiple standards bodies such as the ITU (International Telecommunications Union) and the IETF (Internet Engineering Task Force). Vocoders are another area of continued standards evolution for voice over IP. Most vocoders used as part of VoIP solutions were based on existing standards created for digital telephony applications.
Wireline voice-over-packet gateways are used to convert digital voice signals into packets. Digital pulse code modulation (PCM) voice signals taken from SONET facilities are processed by the gateway. Voice channels are converted from their TDM format to digital “channels” and processed to remove any echo. The signal level is adjusted to appropriate levels and compression algorithms are applied to the channels. Following this, each channel is packetized and transmitted to the packet network. Various other optional processes may be needed to perform such functions as the extraction of signaling information, management of tones or detection of voice activity for comfort noise generation.
Signaling protocols used in wireline networks can be categorized by specific market segments. Service providers predominantly use session initiation protocol (SIP) as the transport protocol of choice. In this application, SIP is often used to replace and/or convert SS7 signaling. Cable operators have favored multimedia gateway control protocol (MGCP) for edge to client signaling. On the enterprise side, the H.323 protocol suite has been deployed almost exclusively.
Codecs follow a similar division based on market segments. ILECs and other large carriers typically focus on G.711 (or traditional PCM) and G.726 (adaptive PCM) due to their existing circuit-switched environment, and because of the ability to perform tandem coding without signal quality degradation. With any of the other available codecs, network design must consider the effects of tandem coding and quality implications.
Enterprises often use G.729AB and G.723.1A due to their low-bandwidth requirements and association with PC networks using Microsoft NetMeeting. The tradeoff here is that the signal quality is less than in the previously discussed codecs, and the network must be designed to control the number of tandem coding incidences. Additionally, the processing power required for a single channel using these codecs is roughly twice that of a PCM channel.
G.729E and G.728 are the primary codecs used by cable operators. These codecs provide a middle-of-the-road approach, their quality being roughly equal to PCM/ADPCM, but with reduced bandwidth. The approach provides quality suitable for more demanding applications (such as providing music while callers hold), but it is also the most expensive. The processing power required for one channel is roughly 3.5 times that required for a single PCM channel.
Under various protocols, individual media codecs have identifiers, which may be a names or a numeric codes. When a call is attempted, media connections (e.g., voice, video, music) will not be established unless the sending and receiving codec have the same identifier. An exact match is not really necessary since a number of codecs are code-level compatible, i.e. they can successfully decode each other's media stream.
Each code-compatible codec group is called companion group, and any two members of the same companion group are called companion codecs. In a companion group, any codec can successfully decode the output of any other group member. For example, G.729 and G.729B form a companion group, and G.729A and G.729AB form another companion group. These examples are not exhaustive, nor are the companion groups limited to voice codecs only.